Internet DRAFT - draft-akhter-opsawg-perfmon-method

draft-akhter-opsawg-perfmon-method






Network Working Group                                          A. Akhter
Internet-Draft                                             Cisco Systems
Intended status: Standards Track                          March 27, 2012
Expires: September 28, 2012


          Methodology for Network Flow Performance Measurement
               draft-akhter-opsawg-perfmon-method-02.txt

Abstract

   There is a need to be able to quantify and report the performance of
   network applications and the network service in handling user data.
   This performance data provides information essential in validating
   service level agreements, fault isolation as well as early warnings
   of network greater problems.  This document describes a generic
   methodology for calculating metrics related to network based
   applications.  In addition, to the performance metrics, several
   additional information elements are included to help provide greater
   context to the reports.  The measurements use audio/video
   applications as base examples but are not restricted to these class
   of applications.

Status of this Memo

   This Internet-Draft is submitted in full conformance with the
   provisions of BCP 78 and BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF).  Note that other groups may also distribute
   working documents as Internet-Drafts.  The list of current Internet-
   Drafts is at http://datatracker.ietf.org/drafts/current/.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."

   This Internet-Draft will expire on September 28, 2012.

Copyright Notice

   Copyright (c) 2012 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents
   (http://trustee.ietf.org/license-info) in effect on the date of



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   publication of this document.  Please review these documents
   carefully, as they describe your rights and restrictions with respect
   to this document.  Code Components extracted from this document must
   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.


Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  3
   2.  Terminology  . . . . . . . . . . . . . . . . . . . . . . . . .  4
   3.  General Usage  . . . . . . . . . . . . . . . . . . . . . . . .  5
     3.1.  Quality of Service (QoS) Monitoring  . . . . . . . . . . .  5
     3.2.  Service Level Agreemnt (SLA) Validation  . . . . . . . . .  5
     3.3.  Fault Isolation and Troubleshooting  . . . . . . . . . . .  5
   4.  New Information Elements . . . . . . . . . . . . . . . . . . .  6
     4.1.  Transport Layer  . . . . . . . . . . . . . . . . . . . . .  6
       4.1.1.  perfPacketLoss . . . . . . . . . . . . . . . . . . . .  6
       4.1.2.  perfPacketExpected . . . . . . . . . . . . . . . . . .  8
       4.1.3.  perfPacketLossRate . . . . . . . . . . . . . . . . . .  9
       4.1.4.  perfPacketLossEvent  . . . . . . . . . . . . . . . . . 10
       4.1.5.  perfPacketInterArrivalJitterAvg  . . . . . . . . . . . 11
       4.1.6.  perfPacketInterArrivalJitterMin  . . . . . . . . . . . 12
       4.1.7.  perfPacketInterArrivalJitterMax  . . . . . . . . . . . 13
       4.1.8.  perfRoundTripNetworkDelay  . . . . . . . . . . . . . . 13
     4.2.  User and Application Layer . . . . . . . . . . . . . . . . 14
       4.2.1.  perfSessionSetupDelay  . . . . . . . . . . . . . . . . 14
     4.3.  Contextual Elements  . . . . . . . . . . . . . . . . . . . 15
       4.3.1.  mediaRTPSSRC . . . . . . . . . . . . . . . . . . . . . 15
       4.3.2.  mediaRTPPayloadType  . . . . . . . . . . . . . . . . . 16
       4.3.3.  mimeType . . . . . . . . . . . . . . . . . . . . . . . 16
   5.  Security Considerations  . . . . . . . . . . . . . . . . . . . 17
   6.  Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 17
   7.  References . . . . . . . . . . . . . . . . . . . . . . . . . . 18
     7.1.  Normative References . . . . . . . . . . . . . . . . . . . 18
     7.2.  Informative References . . . . . . . . . . . . . . . . . . 18
   Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 20













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1.  Introduction

   Today's networks support a multitude of highly demanding and
   sensitive network applications.  Network issues are readily apparent
   by the users of these applications due to the sensitivity of these
   applications to impaired network conditions.  Examples of these
   network applications include applications making use of IP based
   audio, video, database transactions, virtual desktop interface (VDI),
   online gaming, cloud services and many more.  In some cases, the
   impaired application translates directly to loss of revenue.  In
   other cases, there may be regulatory or contractual service level
   agreements that motivate the network operator.  Due to the
   sensitivity of these types of applications to impaired service, it
   leaves a poor impression of the network service on the user--
   regardless of the actual performance of the network itself.  In the
   case of an actual problem within the network service, monitoring the
   performance may yield an early indicator of a much more serious
   problem.

   Due to the demanding and sensitive nature of these applications,
   network operators have tried to engineer their networks towards
   wringing better and differentiated performance.  However, that same
   differentiated design prevents network operators from extrapolating
   observational data from one application to another, or from one set
   of synthetic (active test) test traffic to actual application
   performance.  This gap highlights the importance of generic
   measurements as well as the reliance on user traffic measurents--
   rather than synthetic tests.

   Performance measurements on user data provide greater visibility not
   only into the quality of experience of the end users but also
   visibility into network health.  With regards to network health, as
   flow performance is being measured, there will be visibility into the
   end to end performance which means that not only visibility into
   local network health, but also viability into remote network health.
   If these measurements are made at multiple points within the network
   (or between the network and end device) then there is not only
   identification that there might be an issue, but a span of area can
   be established where the issue might be.  The resolution of the fault
   increases with the number of measurement points along the flow path.

   The IP Flow Information Export Protocol (IPFIX) [RFC5101] provides
   new levels of flexibility in reporting from measurement points across
   the life cycle of a network based application.  IPFIX can provide
   granular results in terms of flow specificity as well as time
   granularity.  At the same time, IPFIX allows for summarization of
   data along different types of boundaries for operators that are
   unconcerned about specific sessions but about health of a service or



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   a portion of the network.  This document details the methodlogy of
   measurement, while an accompanying document describes the expression
   of the measurements in IPFIX format.

   Where possible, an attempt has been made to make use of existing
   definitions of metrics ([RFC4710]) and if needed, clarify and expand
   on them to widen their usage with additional applications, and
   network devices.  For example, the RTP measurments have generally
   defined from the prespective of end systems rather than intermdiate
   nodes which are not alwyas privy to the application context and may
   have limited scaling properties.  The methodology described in
   [I-D.ietf-pmol-sip-perf-metrics] is used to describe the methodology
   of measurement.

   There has been related work in this area such as [RFC2321].
   [I-D.huici-ipfix-sipfix], and [VoIP-monitor].  This document is also
   an attempt to generalize as well as standardize the reporting formats
   and measurement methodology.


2.  Terminology

   Terms used in this document that are defined in the Terminology
   section of the IPFIX Protocol [RFC5101] document are to be
   interpreted as defined there.

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
   document are to be interpreted as described in RFC 2119 [RFC2119].

   In addition, the information element definitions use the following
   terms:

   Name:  Name of the information element

   Description:  Short description of what the information element is
      trying to convey.

   Observation Point:  Where the measurement is meant to be performed.
      Either at an intermediate system (for example, a router) or end
      system.

   Use and Applications  An explanation of how this particular
      information element would be used.







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   Calculation Method:  In the case of metrics, this section describes
      how the metric is calculated, as well as any special conditions.

   Units of Measurement:  In the case of metrics, what are the units of
      measurement.  The text here is expected to be wider and more
      descriptive than in the IPFIX Element Units section.

   Measurement Timing:  Discussion on the acceptable range of timing and
      sampling intervals.


3.  General Usage

3.1.  Quality of Service (QoS) Monitoring

   The network operator needs to be able to gauge the end user's
   satisfaction with the network service.  While there are many
   components of the satisfaction such as pricing, packaging, offering,
   etc., a major component of satisfaction is delivering a consistent
   service.  The user builds trust on this consistency of the network
   service and runs network applications with confidence-- which is of
   course the end goal.  Without the ability to deliver a consistent
   service for end user network applications network operator will be
   left dealing with price sensitive disgruntled users with very low
   expectations and utilization (if they don't have choice of operator)
   or abandonment (if they have choice).

3.2.  Service Level Agreemnt (SLA) Validation

   Similar to QoS and QoE validation, there might be contractual or
   regulatory requirements that need to be met by the network operator.
   Monitoring the performance of the flows allows the application
   operator, network operator as well as the end user to validate if the
   target service is being delivered.  While there is quite a diversity
   in the codification of network SLAs, the eventually involve some
   measurement of network uptime, end to end latency, end to end jitter
   and perhaps service response time.  In the case of SLA violation, the
   start and end times, nature and network scope of the violation needs
   to be captured to allow for the most accurate settling of the SLA
   violation.

3.3.  Fault Isolation and Troubleshooting

   It has been generally easier to troubleshoot and fix problems that
   are binary in nature: it either works or does not work.  The host is
   pingable or not pingable.  However, it is the much more difficult to
   resolve transitory issues that move from location to location, are
   not complete faiures and sometimes with unverfifiable end user



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   reports as the only indication of a problem.  It is these
   intermittent and seemingly inconsistent network impairments that
   performance metrics can be extremely helpful with.  Just the basic
   timely detection that there is a problem (or an impending problem)
   can give the operator provider the confidence that there is a real
   problem that needs to be resolved.  The next step would be to assist
   the operator in a speedy resolution by providing information
   regarding the network location and nature of the problem.


4.  New Information Elements

   The information elements are organized into two main groups:

   Transport Layer:  Metrics that might be calculated from observations
      at higher layers but essentially provide information about the
      network transport of user date.  For example, the metrics related
      to packet loss, latency and jitter would be defined here.

   User and Application Layer:  Metrics that are might be affected by
      the network indirectly, but are ultimately related to user, end-
      system and session states.  For example, session setup time,
      transaction rate and session duration would be defined here.

   Contextual Elements:  Information elements that provide further
      context to the metrics.  For example, media type, codec type, and
      type of application would be defined here.

4.1.  Transport Layer

4.1.1.  perfPacketLoss

   Name:  perfPacketLoss

   Description:  The packet loss metric reports the number of individual
      packets that were lost in the reporting interval.

   Observation Point:  The observation can be made anywhere along the
      media path or on the endpoints them selves.  The observation is
      only relevant in a unidirectional sense.

   Use and Applications  The packet loss metric can be used to determine
      if there is a network impairment that is causing packet loss
      upstream of the measurement point.  When there are observation
      points on either side of the impairment location it is possible to
      locate the impairment.  With the location information the operator
      can is able to perform quicker fault-isolation as well as shorten
      time to resolution.  Depending on implementation and operator



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      configuration, the granularity of contextual information can be
      very specific.  For example, these traffic loss statistics when
      sent with IP subnet or DSCP information can provide visibilty into
      QoS specific or network topology issues.

   Calculation Method:  This metric requires that each IP packet be
      individually marked with a monotonically incrementing sequence
      number.  A number of encapsulations support this type of
      sequencing: IPSec ESP [RFC4303], GRE [RFC2890] and RTP [RFC3550].
      An analysis of the sequence number field can yield the lost number
      of packets.  In certain cases, there might be an element of
      discovery and synchronization of the flow itself before the
      measurement can be made.  An example of this can be found for RTP
      flows running on ephemeral UDP port numbers.  In these cases,
      reporting 0 as packet loss would be misleading and the value
      0xFFFFFFFF MUST be used in cases where the packet loss value
      cannot be determined.  In the case of a monitor interval where
      synchronization was achieved mid-interval, the loss packet counter
      MAY be used to represent the remainder of the interval.  As this
      metric is a deltaCounter, the number of loss packets only
      represent the observation within the reporting interval.  Due to
      the dependency on the arrival of a packet with a sequence number
      to calculate loss, the loss calculation may be indefinitely
      delayed if no more packets arrive at all.  For the case of RTP, in
      addition to the 16 bit sequence number field in RFC3550, there is
      also the additional 16-bit high-order sequence number field (for a
      total of 32-bit seq number space) that is used in RFC3497
      [RFC3497].  RFC3497 traffic runs at a very high rate and the 32-
      bit field allow for additional time for wrapping (21 seconds).
      So, a loss span of greater than 21 seconds measured only by the
      16-bit field will lead to inaccurate reporting.  In the case of
      secure RTP [RFC3711], the relevant portion of the RTP header is in
      the clear and lost packet counting can still be performed.  It is
      important to note that the sequence number space is unique per RTP
      SSRC.  Therefore it is important to track the high sequence number
      seen on a per SSRC-5-tuple basis.  There may be multiple SSRCs in
      a single 5-tuple.  Certain applications inject non-RTP traffic
      into the same 5-tuple as the media stream.  RTCP packets may be
      seen in the same 5-tuple as the RTP stream [RFC5761], and STUN
      [RFC5389] packets may also be seen.  The loss detection should
      ignore these packets.  There may be spans within the network where
      header compression schemes such as [RFC2508] are used.  In cases
      where the measurement device is terminating the compression, and
      the measurement implementation does not support calculation of the
      metric the value 0xFFFFFFFF MUST be reported.  In other cases the
      measurement point may be at a midpoint of the header compression
      network span.  Depending on the mechanics of header compression,
      sequencing information may be present and it is possible to



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      calculate the metric.  In such cases the implementation SHOULD
      perform the calculation and report the metric.

   Units of Measurement:  packets

   Measurment Timing  To be able to calculate this metric a continuous
      set of the flow's packets (as each would have an incrementing
      sequence number) needs to be monitored.  Therefore, per-packet
      sampling would prevent this metric from being calculated.
      However, there are other sampling methodologies that might be
      usable.  It is possible to generate sampled metrics by sampling
      spans of continuous packets, however a portion of the span may
      have to be utilized for resynchronization of the sequence number.
      Another form of acceptable sampling would be at the flow level.

4.1.2.  perfPacketExpected

   Name:  perfPacketExpected

   Description:  The number of packets there were expected within a
      monitoring interval.

   Observation Point:  The observation can be made anywhere along the
      media path or on the endpoints them selves.  The observation is
      only relevant in a unidirectional sense.

   Use and Applications  The perfPacketExpected is a mid-calculation
      metric used in the generation of perfPacketLossRate.  It is
      equivilent to the highest received packet sequence number at the
      time of measurement.  As the value only increments when packets
      are received, packet loss may be occouring at the time of
      measurement but perfPacketExpected remains constant.

   Calculation Method:  The subtraction of the last sequence number from
      the first sequence number in monitoring interval yields the
      expected count.  As discussed with perfPacketLost, there might be
      a delay due to synchronization with the flow's sequence numbers
      and in such times the value of the metric should be set to
      0xFFFFFFFE.  Care has to be taken to account for cases where the
      packet's sequence number field wraps.  For RTP, the expected count
      calculation formula can be found in Appendix A.3 of [RFC3550].
      Refer to the perfPacketLoss metric regarding considerations for
      header compression.  The value 0xFFFF is used to represent cases
      where the metric could not be calculated.







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   Units of Measurement:  packets

   Measurment Timing  Same considerations as perfPacketLoss

4.1.3.  perfPacketLossRate

   Name:  perfPacketLossPercentage

   Description:  Percentage of number of packets lost out of the total
      set of packets sent.

   Observation Point:  The observation can be made anywhere along the
      media path or on the endpoints them selves.  The observation is
      only relevant in a unidirectional sense.

   Use and Applications  The perfPacketLossRate metric can be used to
      normalize the perfPacketLoss metric to handle cases where
      different flows are running at different packet per second (PPS)
      rates.  Due to the normalization, comparisons can now be made
      against thresholds (for creating alerts, etc.).  In addition, the
      percentage form of the metric allows for comparisons against other
      flows at the same observation point to determine if there is an
      equal bias for drops between the flows.  Otherwise, the
      perfoPacketLossRate is used in same way as perfPacketLoss.  This
      value can be derived from perfPacketExpected and perfPacketLoss
      and is offered as a convenience to ease functions such as
      thresholding, and pre-computed reporting.  It shoudl be noted that
      for large values of perfPacketExpected and perfPacketLoss it might
      be preferable and more accurate for the conversion to percentage
      to occour at a later stage where the accuracy can be controlled.

   Calculation Method:  The number of lost packets divided by the number
      of expected packets in an interval period multiplied by 100.  In
      cases where perfPacketLoss is unknown (for example due to
      synchronization issues), the perfPacketLossRate would also be
      unknown.  If there are multiple flows whose loss rate is being
      aggregated, then the average of the individual flows is used.
      Refer to the perfPacketLoss metric regarding considerations for
      header compression.

   Units of Measurement:  percentage

   Measurment Timing  Same notes as perfPacketLossPercentage








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4.1.4.  perfPacketLossEvent

   Name:  perfPacketLossEvent

   Description:  The packet loss event metric reports the number of
      continuous sets of packets that were lost in the reporting
      interval.

   Observation Point:  The observation can be made anywhere along the
      media path or on the endpoints them selves.  The observation is
      only relevant in a unidirectional sense.

   Use and Applications  The perfPacketLossEvent metric can provide loss
      information for protocols that do not implement per packet
      sequencing.  Similarly to the perfPacketLoss metric, the packet
      loss event metric can be used to determine if there is a network
      impairment that is causing packet loss upstream of the measurement
      point.  In cases where both the perfPacketLoss and
      perfPacketLossEvent metric are available, the ratio between the
      packet loss and packet event count can provide the average loss
      length.  The average loss length provides additional information
      regarding the cause of the loss.  For example, a dirty fiber
      connection might have a low average loss length, while a routing
      protocol convergence will have a high loss length.

   Calculation Method:  This data value is a simplified version of the
      Lost Packets metric.  Whereas Lost Packets counts individual
      packet loss, the 'loss event count' metric counts sets of packets
      that are lost.  For example, in the case of a sequence of packets:
      1,3,6,7,10 the packets marked 2,4,5,8 and 9 are lost.  So, a total
      of 5 packets are lost.  This same sequence translates to 3 loss
      events: (2), (4,5) and (8,9).  In the case of RTP, the sequence
      number in the RTP header can be used to identify loss events.
      Certain protocols such as TCP and UDP+MPEG2-TS encapsulation in IP
      have sequencing information, but the sequence field is incremented
      by individual IP packets.  As a side note, in the case of UDP+
      MPEG2-TS encapsulation the simple use of RTP+MPEG2-TS via
      [RFC2250] results in the avaliability of the more granular
      perfPacketLoss metrics.  In these cases, the perfPacketLoss metric
      cannot be calculated but the perfPacketLossEvent can be calculated
      and can provide detection of loss.  The value 0xFFFFFFFF is used
      to represent non-applicable cases such as lack of sequence number
      synchronization.  Many of the same considerations as for
      perfPacketLoss apply to perfPacketLoss event.  Please refer to the
      Calculation Method section of the perfPacketLoss.






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   Units of Measurement:  event counts

   Measurment Timing  Please refer to the measurement timing section of
      perfPacketLoss.

4.1.5.  perfPacketInterArrivalJitterAvg

   Name:  perfPacketInterArrivalJitterAvg

   Description:  This metric measures the absolute deviation of the
      difference in packet spacing at the measurement point compared to
      the packet spacing at the sender.

   Observation Point:  The observation can be made anywhere along the
      media path or on the receiver.  The observation is only relevant
      in a unidirectional sense.

   Use and Applications  The inter arrival jitter data value can be used
      be network operator to determine the network's impact to the
      spacing in between a media stream's packets as they traverse the
      network.  For example, in the case of media applications, the
      receiving end system is expecting these packets to come in at a
      particular periodicity and large deviations may result in de-
      jitter buffers adding excessive delay, or the media packets being
      discarded.  When the data is reported from multiple intermediate
      nodes, the area of the network that is having a detrimental
      contribution can be identified.  On a non-media application level,
      the inter arrival jitter metrics can be used for early indication
      queuing contention within the network (which could lead to packet
      loss).

   Calculation Method:  The inter arrival jitter value makes use of the
      association of sending time with an IP packets and comparison of
      the arrival time on the monitoring point.  In certain protocols, a
      representation of sending time is encoded into the header itself.
      For example, in the case of RTP packets, the RTP header's
      timestamps field represents encoder clock ticks-- which are
      representations of time.  Similarly, in the case of TCP options
      encode absolute timestamps values.  For RTP the calculation method
      can be found in Appendix A of [RFC3550].  It should be noted that
      the RFC3550 calculation is on the last 16 packets measured.  The
      most recent value calculated SHOULD be reported at the end of the
      monitoring interval.  The range of the jitter values during the
      monitoring interval can be reported using
      perfPacketInterArrivalJitterMin and
      perfPacketInterArrivalJitterMax.  Similarly to the perfPacketLoss
      case there may be periods of time where the jitter value cannot be
      calculated.  In these cases, the 0xFFFFFFFF value should be used



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      to convey the lack of availability of the metric.  As mentioned
      earlier, the RTP header timestamps is actually a 'sample-stamp'
      (ie clicks) from the encoder's clock.  The frequency of the clock
      is dependent on the codec.  Some codecs (eg AAC-LD) support
      multiple possible frequencies one of which is then selected for
      the media-stream.  The mapping to clock rate can be performed via
      mapping from the static RTP payload type (RTP-PT), but newer
      codecs are make use of the dynamic payload type range and the
      RTP-PT (in the dynamic case) cannot be used to determine the clock
      frequency.  There are various methods by which the clock frequency
      (deep packet inspection of the signalling, manual configuration,
      etc.) can be associated to the calculation method.  The frequency
      should be locked in the metering layer to a unique combination of
      the IP source, IP destination, IP protocol layer-4 ports, RTP-PT
      and SSRC.  By strict RFC3550 definition, the SSRC is set to a
      specific encoder clock and it is the SSRC that should be tracked
      rather than payload type.  However, in recent discussions it has
      been noted that there are RTP implementations that might change
      the encoder clock frequency while maintaining the SSRC value.  An
      encoder frequency change will be accompanied by a different
      RTP-PT.

   Units of Measurement:  microseconds

   Measurment Timing  Please refer to the measurement timing section of
      perfPacketLoss.

4.1.6.  perfPacketInterArrivalJitterMin

   Name:  perfPacketInterArrivalJitterMin

   Description:  This metric measures the minimum value the calculation
      used for perfPacketInterArrivalJitterAvg within the monitoring
      interval.

   Observation Point:  The observation can be made anywhere along the
      media path or on the receiver.  The observation is only relevant
      in a unidirectional sense.

   Use and Applications  Please refer to the 'Use and Applications'
      section of perfPacketInterArrivalJitterAvg.  This specific metric,
      along with perfPacketInterArrivalJitterMax, is to capture the
      range of measurements observed within a monitoring interval as the
      average function may hide extremes.







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   Calculation Method:  Please see the perfPacketInterArrivalJitterAvg
      section for general calculation section.  The average calculation
      is evaluated on a running basis over the last 16 packets and the
      entire monitoring interval is not covered.  In this metric, the
      minimum value is taken over the entire monitoring interval.

   Units of Measurement:  microseconds

   Measurment Timing  Please refer to the measurement timing section of
      perfPacketLoss.

4.1.7.  perfPacketInterArrivalJitterMax

   Name:  perfPacketInterArrivalJitterMax

   Description:  This metric measures the maximum value the calculation
      used for perfPacketInterArrivalJitterAvg within the monitoring
      interval.

   Observation Point:  The observation can be made anywhere along the
      media path or on the receiver.  The observation is only relevant
      in a unidirectional sense.

   Use and Applications  Please refer to the 'Use and Applications'
      section of perfPacketInterArrivalJitterAvg.  This specific metric,
      along with perfPacketInterArrivalJitterMin, is to capture the
      range of measurements observed within a monitoring interval as the
      average function may hide extremes.

   Calculation Method:  Please see the perfPacketInterArrivalJitterAvg
      section for general calculation section.  The average calculation
      is evaluated on a running basis over the last 16 packets and the
      entire monitoring interval is not covered.  In this metric, the
      maximum value is taken over the entire monitoring interval.

   Units of Measurement:  microseconds

   Measurment Timing  Please refer to the measurement timing section of
      perfPacketLoss.

4.1.8.  perfRoundTripNetworkDelay

   Name:  perfRoundTripNetworkDelay

   Description:  This metric measures the network round trip time
      between end stations for a flow.





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   Observation Point:  The observation can be made anywhere along the
      flow path as long as the bidirectional network delay is accounted
      for.

   Use and Applications  The perfRoundTripNetworkDelay metric can be
      used in multiple ways.  If the applicaiton being monitored
      provides interactive feedback to the user the
      perfRoundTripNetworkDelay can be used to judge the 'liveliness' of
      the application experience.  Other use cases may involve
      troubleshooting throughput issues where the transport protocol's
      throughtput is affected by network delay.

   Calculation Method:  perfRoundTripNetworkDelay can estimated by
      accounting for the network flight time between a transport
      protocol request and response.  In the case of TCP, this would the
      time difference between the TCP SYN and ACK packets in the TCP
      handshake.  It should be noted that at times other than the TCP
      handshake the time difference between TCP end station packet.  For
      RTP (RFC3550) based applications, the network round trip can be
      calculated by analysis of hte RTCP sending and receive times.

   Units of Measurement:  microseconds

   Measurment Timing  Depending on the method used to calculate the
      round trip time, the measurment may only be possible at specific
      times during the session lifecycle.  In time periods where the
      metric is not current 'not calculated' SHOULD be reported.

4.2.  User and Application Layer

4.2.1.  perfSessionSetupDelay

   Name:  perfSessionSetupDelay

   Description:  The Session Setup Delay metric reports the time taken
      from a request being initiated by a host/endpoint to the response
      (or request indicator) to the request being observed.  This metric
      is defined in [RFC4710], however the units have been updated to
      microseconds.

   Observation Point:  This metric needs to be calculated where both
      request and response can be observed.  This could be at network
      choke points, application proxies, or within the end systems
      themselves.







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   Use and Applications  The session setup delay metric can measure the
      end user initial wait experience as seen from the network
      transaction level.  The value will not only include the network
      flight time, but also includes the server response time and may be
      used to alert the operator in cases where the overall service is
      overloaded and thus sluggish, or within normal operating values.

   Calculation Method:  Measure distance in time between the first bit
      of request and the first bit of the response.  For the case of
      SIP, please see Section 4.3.1 of [I-D.ietf-pmol-sip-perf-metrics]

   Units of Measurement:  microseconds

   Measurment Timing  This measurement can be sampled on a session by
      session basis.  It may be advisable to set sample targets on a per
      source range - to destination basis.  Due to the nature of
      measurement intervals, there may be a period of time (and thus
      measurement reports) in which the perfSessionSetupDelay value has
      not been calculated.  In these cases the value 0xFFFFFFFE MUST be
      used and can be interpreted to mean not applicable.  For
      measurement intervals after perfSessionSetupDelay has been
      calculated and the existing calculated perfSessionSetupDelay value
      SHOULD be sent if reporting only on that single session.  However,
      if multiple sessions are summarized in the report then the average
      for perfSessionSetupDelay values calculated in the most recent
      interval SHOULD be used.  The intention with this behavior is to
      acknowledge that the value has not bee calculated, and when it has
      provide the freshest values available.

4.3.  Contextual Elements

4.3.1.  mediaRTPSSRC

   Name:  mediaRTPSSRC

   Description:  Value of the synchronization source (SSRC) field in the
      RTP header of the flow.  This field is defined in [RFC3550]

   Observation Point:  This metric can be gleaned from the RTP packets
      directly, so the observation point needs to on the flow path or
      within the endpoints.

   Use and Applications  The RTP SSRC value denotes a specific media
      stream.  As such when trying to differentiate media stream
      problems between session participants the SSRC field is needed.






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   Calculation Method:  Copy from RTP header's SSRC field as defined in
      [RFC3550].  In the case of a non-RTP flow, or the time period in
      which the flow has not been verified to be a RTP flow the value
      0xFFFFFFFE MUST be reported.

   Units of Measurement:  identifier

   Measurment Timing  It is possible that the SSRC may have be
      renegotiated mid-session due to collisions with other RTP senders.

4.3.2.  mediaRTPPayloadType

   Name:  mediaRTPPayloadType

   Description:  The value of the RTP Payload Type Field as seen in the
      RTP header of the flow.  This field is defined in [RFC3550]

   Observation Point:  This metric can be gleaned from the RTP packets
      directly, so the observation point needs to on the flow path or
      within the endpoints.

   Use and Applications  The RTP PT conveys the payload format and media
      encoding used in the RTP payload.  For simple cases, where the RTP
      PT is from the statically defined range this can lead to an
      understanding of type of media codec used.  With the knowledge of
      the codec being used the degree of media impairment (given loss
      values and jitter) can be estimated better.  However, for more
      recent codecs, the RTP dynamic range is used.  In these cases the
      RTP payload values are dynamically negotiated.  In the case of a
      non-RTP flow, or the time period in which the flow has not been
      verified to be a RTP flow, the value 0xFFFF MUST be reported.

   Calculation Method:  Copy from RTP header's RTP-PT field as defined
      in [RFC3550]

   Units of Measurement:  identifier

   Measurment Timing

4.3.3.  mimeType

   Name:  mimeType

   Description:  The mime type describes the content of the flow.







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   Observation Point:  The ideal location of this metric is on the
      application generators and consumers.  However, given application
      signalling inspection or static configuration it is possible that
      intermediate nodes are able to generate mime type (eg. codec name)
      information.

   Use and Applications  The mime type value conveys information
      regarding the content of a flow.  For example, in the case of
      Audio/Video applications the name of the codec used to encode the
      media in the flow.  Simply reporting loss and jitter measurements
      are useful for detection of network problems.  However, judging
      the degree of the impact on the audio/video experience needs
      additional information.  The most basic information is the codec
      being used which when coupled with per-codec knowledge of
      sensitivity to the transport metrics a better idea of the
      experience can be gained.

   Calculation Method:  The valid values for the mime type are listed on
      the IANA mime type registry.  For Audio/Video codecs, there is a
      specific media-types registry.  Analysis of the RTP payload type
      may lead to the determination of the media codec.  However, with
      the use of the RTP dynamic payload type range the media
      information is not encoded into the data packet.  For these cases,
      intermediate nodes may need to perform inspection of the
      signalling (SIP, H.323, RTSP, etc.).  In cases where the
      mediaCodec cannot be determined, the value 'unknown' MUST be used.

   Units of Measurement:  identifier

   Measurment Timing


5.  Security Considerations

   The recommendations in this document do not introduce any additional
   security issues to those already mentioned in [RFC5101] and [RFC5477]


6.  Acknowledgements

   The authors would like to thank Rahul Patel, Jan Novak, Al Morton,
   Brad Fawcett, Doug Manley and Shingo Kashima for their invaluable
   review and comments.  Thank-you.


7.  References





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7.1.  Normative References

   [RFC5101]  Claise, B., "Specification of the IP Flow Information
              Export (IPFIX) Protocol for the Exchange of IP Traffic
              Flow Information", RFC 5101, January 2008.

   [RFC5610]  Boschi, E., Trammell, B., Mark, L., and T. Zseby,
              "Exporting Type Information for IP Flow Information Export
              (IPFIX) Information Elements", RFC 5610, July 2009.

   [RFC4710]  Siddiqui, A., Romascanu, D., and E. Golovinsky, "Real-time
              Application Quality-of-Service Monitoring (RAQMON)
              Framework", RFC 4710, October 2006.

   [RFC5102]  Quittek, J., Bryant, S., Claise, B., Aitken, P., and J.
              Meyer, "Information Model for IP Flow Information Export",
              RFC 5102, January 2008.

   [RFC3550]  Schulzrinne, H., Casner, S., Frederick, R., and V.
              Jacobson, "RTP: A Transport Protocol for Real-Time
              Applications", STD 64, RFC 3550, July 2003.

   [RFC3497]  Gharai, L., Perkins, C., Goncher, G., and A. Mankin, "RTP
              Payload Format for Society of Motion Picture and
              Television Engineers (SMPTE) 292M Video", RFC 3497,
              March 2003.

   [RFC5389]  Rosenberg, J., Mahy, R., Matthews, P., and D. Wing,
              "Session Traversal Utilities for NAT (STUN)", RFC 5389,
              October 2008.

   [I-D.ietf-pmol-sip-perf-metrics]
              Malas, D. and A. Morton, "Basic Telephony SIP End-to-End
              Performance Metrics", draft-ietf-pmol-sip-perf-metrics-07
              (work in progress), September 2010.

   [iana-ipfix-assignments]
              Internet Assigned Numbers Authority, "IP Flow Information
              Export Information Elements
              (http://www.iana.org/assignments/ipfix/ipfix.xml)".

7.2.  Informative References

   [I-D.ietf-pmol-metrics-framework]
              Clark, A. and B. Claise, "Guidelines for Considering New
              Performance Metric Development",
              draft-ietf-pmol-metrics-framework-12 (work in progress),
              July 2011.



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   [RFC2508]  Casner, S. and V. Jacobson, "Compressing IP/UDP/RTP
              Headers for Low-Speed Serial Links", RFC 2508,
              February 1999.

   [RFC3711]  Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
              Norrman, "The Secure Real-time Transport Protocol (SRTP)",
              RFC 3711, March 2004.

   [RFC2250]  Hoffman, D., Fernando, G., Goyal, V., and M. Civanlar,
              "RTP Payload Format for MPEG1/MPEG2 Video", RFC 2250,
              January 1998.

   [RFC2890]  Dommety, G., "Key and Sequence Number Extensions to GRE",
              RFC 2890, September 2000.

   [RFC4303]  Kent, S., "IP Encapsulating Security Payload (ESP)",
              RFC 4303, December 2005.

   [RFC5761]  Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
              Control Packets on a Single Port", RFC 5761, April 2010.

   [I-D.huici-ipfix-sipfix]
              Huici, F., Niccolini, S., and S. Anderson, "SIPFIX: Use
              Cases and Problem Statement for VoIP Monitoring and
              Exporting", draft-huici-ipfix-sipfix-00 (work in
              progress), June 2009.

   [nProbe]   "nProbe - NetFlow/IPFIX Network Probe
              (http://www.ntop.org/nProbe.html)".

   [RFC2321]  Bressen, A., "RITA -- The Reliable Internetwork
              Troubleshooting Agent", RFC 2321, April 1998.

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119, March 1997.

   [RFC5477]  Dietz, T., Claise, B., Aitken, P., Dressler, F., and G.
              Carle, "Information Model for Packet Sampling Exports",
              RFC 5477, March 2009.

   [VoIP-monitor]
              L. Chang-Yong, H. Kim, K. Ko, J. Jim, and H. Jeong, "A
              VoIP Traffic Monitoring System based on NetFlow v9,
              International Journal of Advanced Science and Technology,
              vol. 4, Mar. 2009".






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Author's Address

   Aamer Akhter
   Cisco Systems, Inc.
   7025 Kit Creek Road
   RTP, NC  27709
   USA

   Email: aakhter@cisco.com










































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